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bobg at techie.com
Guest
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 Posted: Thu Feb 28, 2008 10:51 pm Post subject: [asterisk-biz] Ribbit.com ? 1ezphone.com |
 |
|
Try http://1ezphone.com
| Quote: | ----- Original Message -----
From: "Mike Clark"
To: email@mattruby.com, "Commercial and Business-Oriented Asterisk Discussion"
Subject: Re: [asterisk-biz] Ribbit.com ?
Date: Mon, 17 Dec 2007 17:21:50 -0500
Matthew Rubenstein wrote:
| Quote: | Dean, how would you describe Mexuar, with its embeddable but
proprietary IAX applet, in that context?
|
...snipped a bunch..
I'm not Dean, but I'll comment here.I evaluated Mexuar and really liked
it, but they had no good mechanism for a small developer to get started.
They wanted a substantial up front licensing fee to get going. OTOH, if
you turned out to be successful, it was a good deal because it was a one
time fee.
Ribbit has a totally different model as they are a full blown ITSP and
have provided a Flex/Actionscript API to their Flash phone component at
no charge to developers. I have an app ready to roll as soon as they are
completely live.
I would love to see a similar type API to a Flash SIP or IAX2 component
where I could access my own Asterisk or Freeswitch server.
Mike Clark
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asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
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sip at arcdiv.com
Guest
|
| Posted: Fri Feb 29, 2008 7:37 am Post subject: [asterisk-biz] Ribbit.com ? 1ezphone.com |
|
|
I don't know if there's just no real content on that site or if it just
doesn't work with any browser I tried (FF, IE, Opera). Links to any of
the pages other than the 'Enter' page go nowhere, open no information,
etc. Not even a popup warning from ol' FF.
N.
Bob Gibson wrote:
| Quote: | Try http://1ezphone.com
----- Original Message -----
From: "Mike Clark"
To: email@mattruby.com, "Commercial and Business-Oriented Asterisk
Discussion"
Subject: Re: [asterisk-biz] Ribbit.com ?
Date: Mon, 17 Dec 2007 17:21:50 -0500
Matthew Rubenstein wrote:
| Quote: | Dean, how would you describe Mexuar, with its embeddable but
proprietary IAX applet, in that context?
|
...snipped a bunch..
I'm not Dean, but I'll comment here.I evaluated Mexuar and really
liked
it, but they had no good mechanism for a small developer to get
started.
They wanted a substantial up front licensing fee to get going.
OTOH, if
you turned out to be successful, it was a good deal because it was
a one
time fee.
Ribbit has a totally different model as they are a full blown ITSP and
have provided a Flex/Actionscript API to their Flash phone
component at
no charge to developers. I have an app ready to roll as soon as
they are
completely live.
I would love to see a similar type API to a Flash SIP or IAX2
component
where I could access my own Asterisk or Freeswitch server.
Mike Clark
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
--
Want an e-mail address like mine?
Get a *free e-mail *account today at www.mail.com
<http://www.mail.com/Product.aspx>!
------------------------------------------------------------------------
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asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
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|
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trixter at 0xdecafbad.com
Guest
|
| Posted: Fri Feb 29, 2008 7:59 am Post subject: [asterisk-biz] Ribbit.com ? 1ezphone.com |
|
|
| Quote: | | Quote: | ----- Original Message -----
From: "Mike Clark"
To: email@mattruby.com, "Commercial and Business-Oriented Asterisk
Discussion"
Subject: Re: [asterisk-biz] Ribbit.com ?
Date: Mon, 17 Dec 2007 17:21:50 -0500
|
|
| Quote: | | Quote: | Ribbit has a totally different model as they are a full blown ITSP and
have provided a Flex/Actionscript API to their Flash phone
component at
no charge to developers. I have an app ready to roll as soon as
they are
completely live.
I would love to see a similar type API to a Flash SIP or IAX2
component
where I could access my own Asterisk or Freeswitch server.
|
|
Flash does not afaik support UDP so the RTP part would be difficult at
best. I am unsure if the really new versions do or not. Granted you
could have a plugin (flash does have the ability to execute programs
that are in a special directory) which really only would need to be a
tcp->udp converter if you wanted, although it could be a full RTP stack
as well instead of doing that in flash.
Gizmophone has a web component that transmits the audio via HTTPS via
flash. I havent looked at ribbit so I dont know if that is how they are
doing it or not. They also use a plugin to try to limit how many calls
you can do at one time off one box (they did give away free minutes at
one point, they may still do that).
While the SIP RFC requires TCP support for signalling, the media would
still be udp and still be the problem. And if you want to connect to
asterisk you have to use UDP signalling since asterisk does not yet
officially support TCP, despite the RFCs requirement.
Personally what I think would be better is a very simple app that can
send events (on/off hook, dnd/presence, dtmf digits, number dialed, etc)
as well as media (just stream it from the mic direct, which is something
that flash has built in). This would connect to some server side
process that will then connect to whatever protocol you prefer for
termination elsewhere.
On lossy networks you would have a problem of a dropped packet causing a
retransmit, however this may not be that big of a problem in many
environments. If you have any sort of jitter buffer you should be able
to resync the call dynamically so that packet loss does not cause a
growing skew between leg A and leg B. This is probably the biggest
problem to solve, and I do not know how big of a problem it will be for
most users (for some it will be a killer).
Now if they have java installed as well, flash can do liveconnect calls
to the JRE, but if you are going to go that route, it might be better to
just do it all in java to begin with.
Now flash recently aquired a key person that was involved in SIP stuff.
The theory (and some statements officially) indicate that the intention
is to build a proper sip stack into flash, but that has yet to be
released.
There are other bridges that exist to basically do the tcp->udp
translation, which could be run on the users system. Examples include
http://www.transmote.com/flosc/
While this is designed to do the open sound protocol, it would not be
difficult to make it do something else, and if you really know action
script you can get around little things like you dont have to do xml
with the xmlsocket, you can bypass the null byte terminator that is
often sent, etc.
For what is needed to do the tcp->udp bridge it wouldnt be hard to write
that on your own, and then go nuts.
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz |
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charles at mexuar.com
Guest
|
| Posted: Fri Feb 29, 2008 10:56 am Post subject: [asterisk-biz] Ribbit.com ? 1ezphone.com |
|
|
Hello List,
I would like to clarify the way we do business at Mexuar as there was a
snippet of info about us charging a hefty licence fee for Corraleta SDK. For
what you get i.e. a working solution it's not hefty but a fair price.
Do you know you can also try before you buy? We're prepared to cut a 30 day
licence - for free. I don't want to spend my lifetime doing this for free as
I have to feed my children, but as you need some help to get going, let us
know.
If you need help with consultancy and building a solution for your client we
can also do this - but not for free!
Finally, I have just started a Mexuar blog if anyone is interested in taking
a look we will start adding some of our customer's apps. You can find this
at http://mexuarapps.blogspot.com
Thank you.
Regards,
Charles.
Charles Woods
Charles (at) Mexuar (dot) com
Mexuar Communications Ltd
Tel +44 (0)161 866 0028
Mobile +44 (0)7894 338260
www.mexuar.com
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
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bobg at techie.com
Guest
|
| Posted: Sat Mar 01, 2008 12:45 am Post subject: [asterisk-biz] Ribbit.com ? 1ezphone.com |
|
|
Its just prototype click on phone, then register a login and name then click profile to enter IP. login and password for your system.
Or email me and I will setup profile for to make a few calls.
Its no big deal just a softphone but it uses any browser and littler faster protocol.
Ted
ted@1ezphone.com (ted@1ezphone.com)
| Quote: | ----- Original Message -----
From: SIP
To: "Commercial and Business-Oriented Asterisk Discussion"
Subject: Re: [asterisk-biz] Ribbit.com ? 1ezphone.com
Date: Fri, 29 Feb 2008 07:26:12 -0500
I don't know if there's just no real content on that site or if it just
doesn't work with any browser I tried (FF, IE, Opera). Links to any of
the pages other than the 'Enter' page go nowhere, open no information,
etc. Not even a popup warning from ol' FF.
N.
Bob Gibson wrote:
| Quote: | Try http://1ezphone.com
----- Original Message -----
From: "Mike Clark"
To: email@mattruby.com, "Commercial and Business-Oriented Asterisk
Discussion"
Subject: Re: [asterisk-biz] Ribbit.com ?
Date: Mon, 17 Dec 2007 17:21:50 -0500
Matthew Rubenstein wrote:
| Quote: | Dean, how would you describe Mexuar, with its embeddable but
proprietary IAX applet, in that context?
|
...snipped a bunch..
I'm not Dean, but I'll comment here.I evaluated Mexuar and really
liked
it, but they had no good mechanism for a small developer to get
started.
They wanted a substantial up front licensing fee to get going.
OTOH, if
you turned out to be successful, it was a good deal because it was
a one
time fee.
Ribbit has a totally different model as they are a full blown ITSP and
have provided a Flex/Actionscript API to their Flash phone
component at
no charge to developers. I have an app ready to roll as soon as
they are
completely live.
I would love to see a similar type API to a Flash SIP or IAX2
component
where I could access my own Asterisk or Freeswitch server.
Mike Clark
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
-- Want an e-mail address like mine?
Get a *free e-mail *account today at www.mail.com
!
------------------------------------------------------------------------
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
|
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
|
-- Want an e-mail address like mine? [/b]
Get a free e-mail account today at www.mail.com! |
|
| Back to top |
|
|
thp at westhawk.co.uk
Guest
|
| Posted: Tue Mar 04, 2008 3:51 am Post subject: [asterisk-biz] Ribbit.com ? 1ezphone.com |
|
|
(Sorry about the top posting - It's just the way Zimbra does it)
There are a couple of things to look out for here
(straying into tech issues):
1) buffering - TCP tends to get buffered in the kernel to a
much greater extent than udp - so you can easily find yourself
with seconds of latency.
2) codecs - The only low-latency codec supported by flash
is patented and expensive to license, so the gateway to PSTN
or 'normal' VoIP will always have to carry an aditional cost
of the nelly-moser codec license.
3) protocols - Flash is using a streaming protocol (RTSP),
which isn't a VoIP protocol, so has not got the VoIP features
we have come to expect.
All of which is why adobe is (supposed to be) adding SIP
to some future version of Flash.
- Ok, I admit it, I'm biased, I'm in the Java - IAX camp 
But in general I'm sure that this sort of web-telephony
integration is inevitable. See http://www.phonefromhere.com for our
latest experiment - an iGoogle 'phone home' gadget.
Tim.
----- Original Message -----
From: "Trixter aka Bret McDanel" <trixter@0xdecafbad.com>
To: "Commercial and Business-Oriented Asterisk Discussion" <asterisk-biz@lists.digium.com>
Sent: 29 February 2008 12:47:21 o'clock (GMT) Europe/London
Subject: Re: [asterisk-biz] Ribbit.com ? 1ezphone.com
| Quote: | | Quote: | ----- Original Message -----
From: "Mike Clark"
To: email@mattruby.com, "Commercial and Business-Oriented Asterisk
Discussion"
Subject: Re: [asterisk-biz] Ribbit.com ?
Date: Mon, 17 Dec 2007 17:21:50 -0500
|
|
| Quote: | | Quote: | Ribbit has a totally different model as they are a full blown ITSP and
have provided a Flex/Actionscript API to their Flash phone
component at
no charge to developers. I have an app ready to roll as soon as
they are
completely live.
I would love to see a similar type API to a Flash SIP or IAX2
component
where I could access my own Asterisk or Freeswitch server.
|
|
Flash does not afaik support UDP so the RTP part would be difficult at
best. I am unsure if the really new versions do or not. Granted you
could have a plugin (flash does have the ability to execute programs
that are in a special directory) which really only would need to be a
tcp->udp converter if you wanted, although it could be a full RTP stack
as well instead of doing that in flash.
Gizmophone has a web component that transmits the audio via HTTPS via
flash. I havent looked at ribbit so I dont know if that is how they are
doing it or not. They also use a plugin to try to limit how many calls
you can do at one time off one box (they did give away free minutes at
one point, they may still do that).
While the SIP RFC requires TCP support for signalling, the media would
still be udp and still be the problem. And if you want to connect to
asterisk you have to use UDP signalling since asterisk does not yet
officially support TCP, despite the RFCs requirement.
Personally what I think would be better is a very simple app that can
send events (on/off hook, dnd/presence, dtmf digits, number dialed, etc)
as well as media (just stream it from the mic direct, which is something
that flash has built in). This would connect to some server side
process that will then connect to whatever protocol you prefer for
termination elsewhere.
On lossy networks you would have a problem of a dropped packet causing a
retransmit, however this may not be that big of a problem in many
environments. If you have any sort of jitter buffer you should be able
to resync the call dynamically so that packet loss does not cause a
growing skew between leg A and leg B. This is probably the biggest
problem to solve, and I do not know how big of a problem it will be for
most users (for some it will be a killer).
Now if they have java installed as well, flash can do liveconnect calls
to the JRE, but if you are going to go that route, it might be better to
just do it all in java to begin with.
Now flash recently aquired a key person that was involved in SIP stuff.
The theory (and some statements officially) indicate that the intention
is to build a proper sip stack into flash, but that has yet to be
released.
There are other bridges that exist to basically do the tcp->udp
translation, which could be run on the users system. Examples include
http://www.transmote.com/flosc/
While this is designed to do the open sound protocol, it would not be
difficult to make it do something else, and if you really know action
script you can get around little things like you dont have to do xml
with the xmlsocket, you can bypass the null byte terminator that is
often sent, etc.
For what is needed to do the tcp->udp bridge it wouldnt be hard to write
that on your own, and then go nuts.
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz |
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Dean at cognation.net
Guest
|
| Posted: Tue Mar 04, 2008 8:06 am Post subject: [asterisk-biz] Ribbit.com ? 1ezphone.com |
|
|
Well I suppose now is as good a time as any to break cover J
If you are interested in a SIP browser based solution check out;
www.Surphone.com
Yes there are server based and ASP based pricing models, yes it uses Flash – not it doesn’t use Adobe FMS.
No this isn’t related to the work I did with Mexuar, Yes I am consulting with Surphone for the commercialization of their technology and yes I will be selling the technology here in the USA.
If you are a USA based client and have an interest send me an email for more details.
Regards,
Dean Collins
Cognation Pty Ltd
dean@cognation.net
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial).
| Quote: | -----Original Message-----
From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-
bounces@lists.digium.com] On Behalf Of Tim H. Panton
Sent: Tuesday, 4 March 2008 3:46 AM
To: trixter@0xdecafbad.com; Commercial and Business-Oriented Asterisk
Discussion
Subject: Re: [asterisk-biz] Ribbit.com ? 1ezphone.com
(Sorry about the top posting - It's just the way Zimbra does it)
There are a couple of things to look out for here
(straying into tech issues):
1) buffering - TCP tends to get buffered in the kernel to a
much greater extent than udp - so you can easily find yourself
with seconds of latency.
2) codecs - The only low-latency codec supported by flash
is patented and expensive to license, so the gateway to PSTN
or 'normal' VoIP will always have to carry an aditional cost
of the nelly-moser codec license.
3) protocols - Flash is using a streaming protocol (RTSP),
which isn't a VoIP protocol, so has not got the VoIP features
we have come to expect.
All of which is why adobe is (supposed to be) adding SIP
to some future version of Flash.
- Ok, I admit it, I'm biased, I'm in the Java - IAX camp
But in general I'm sure that this sort of web-telephony
integration is inevitable. See http://www.phonefromhere.com for our
latest experiment - an iGoogle 'phone home' gadget.
Tim.
----- Original Message -----
From: "Trixter aka Bret McDanel" <trixter@0xdecafbad.com>
To: "Commercial and Business-Oriented Asterisk Discussion" <asterisk-
biz@lists.digium.com>
Sent: 29 February 2008 12:47:21 o'clock (GMT) Europe/London
Subject: Re: [asterisk-biz] Ribbit.com ? 1ezphone.com
| Quote: | | Quote: | ----- Original Message -----
From: "Mike Clark"
To: email@mattruby.com, "Commercial and Business-Oriented Asterisk
Discussion"
Subject: Re: [asterisk-biz] Ribbit.com ?
Date: Mon, 17 Dec 2007 17:21:50 -0500
|
|
| Quote: | | Quote: | Ribbit has a totally different model as they are a full blown ITSP and
have provided a Flex/Actionscript API to their Flash phone
component at
no charge to developers. I have an app ready to roll as soon as
they are
completely live.
I would love to see a similar type API to a Flash SIP or IAX2
component
where I could access my own Asterisk or Freeswitch server.
|
|
Flash does not afaik support UDP so the RTP part would be difficult at
best. I am unsure if the really new versions do or not. Granted you
could have a plugin (flash does have the ability to execute programs
that are in a special directory) which really only would need to be a
tcp->udp converter if you wanted, although it could be a full RTP stack
as well instead of doing that in flash.
Gizmophone has a web component that transmits the audio via HTTPS via
flash. I havent looked at ribbit so I dont know if that is how they are
doing it or not. They also use a plugin to try to limit how many calls
you can do at one time off one box (they did give away free minutes at
one point, they may still do that).
While the SIP RFC requires TCP support for signalling, the media would
still be udp and still be the problem. And if you want to connect to
asterisk you have to use UDP signalling since asterisk does not yet
officially support TCP, despite the RFCs requirement.
Personally what I think would be better is a very simple app that can
send events (on/off hook, dnd/presence, dtmf digits, number dialed, etc)
as well as media (just stream it from the mic direct, which is something
that flash has built in). This would connect to some server side
process that will then connect to whatever protocol you prefer for
termination elsewhere.
On lossy networks you would have a problem of a dropped packet causing a
retransmit, however this may not be that big of a problem in many
environments. If you have any sort of jitter buffer you should be able
to resync the call dynamically so that packet loss does not cause a
growing skew between leg A and leg B. This is probably the biggest
problem to solve, and I do not know how big of a problem it will be for
most users (for some it will be a killer).
Now if they have java installed as well, flash can do liveconnect calls
to the JRE, but if you are going to go that route, it might be better to
just do it all in java to begin with.
Now flash recently aquired a key person that was involved in SIP stuff.
The theory (and some statements officially) indicate that the intention
is to build a proper sip stack into flash, but that has yet to be
released.
There are other bridges that exist to basically do the tcp->udp
translation, which could be run on the users system. Examples include
http://www.transmote.com/flosc/
While this is designed to do the open sound protocol, it would not be
difficult to make it do something else, and if you really know action
script you can get around little things like you dont have to do xml
with the xmlsocket, you can bypass the null byte terminator that is
often sent, etc.
For what is needed to do the tcp->udp bridge it wouldnt be hard to write
that on your own, and then go nuts.
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
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